Implements AudioMessageFilter as member in RenderThread

content/renderer/media/audio_device.cc

 // Copyright (c) 2011 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/renderer/media/audio_device.h"
 
-#include "base/memory/singleton.h"
 #include "base/message_loop.h"
 #include "content/common/child_process.h"
 #include "content/common/media/audio_messages.h"
 #include "content/common/view_messages.h"
 #include "content/renderer/render_thread.h"
 #include "media/audio/audio_util.h"
 
-scoped_refptr<AudioMessageFilter> AudioDevice::filter_;
-
-namespace {
-
-// AudioMessageFilterCreator is intended to be used as a singleton so we can
-// get access to a shared AudioMessageFilter.
-// Example usage:
-//   AudioMessageFilter* filter = AudioMessageFilterCreator::SharedFilter();
-
-class AudioMessageFilterCreator {
- public:
-  AudioMessageFilterCreator() {
-    int routing_id;
-    RenderThread::current()->Send(
-        new ViewHostMsg_GenerateRoutingID(&routing_id));
-    filter_ = new AudioMessageFilter(routing_id);
-    RenderThread::current()->AddFilter(filter_);
-  }
-
-  static AudioMessageFilter* SharedFilter() {
-    return GetInstance()->filter_.get();
-  }
-
-  static AudioMessageFilterCreator* GetInstance() {
-    return Singleton<AudioMessageFilterCreator>::get();
-  }
-
- private:
-  scoped_refptr<AudioMessageFilter> filter_;
-};
-
-}  // namespace
-
 AudioDevice::AudioDevice(size_t buffer_size,
                          int channels,
                          double sample_rate,
                          RenderCallback* callback)
     : buffer_size_(buffer_size),
       channels_(channels),
       bits_per_sample_(16),
       sample_rate_(sample_rate),
       callback_(callback),
       audio_delay_milliseconds_(0),
       volume_(1.0),
       stream_id_(0) {
+  filter_ = RenderThread::current()->audio_message_filter();
   audio_data_.reserve(channels);
   for (int i = 0; i < channels; ++i) {
     float* channel_data = new float[buffer_size];
     audio_data_.push_back(channel_data);
   }
-  // Lazily create the message filter and share across AudioDevice instances.
-  filter_ = AudioMessageFilterCreator::SharedFilter();
 }
 
 AudioDevice::~AudioDevice() {
   // Make sure we have been shut down.
   DCHECK_EQ(0, stream_id_);
   Stop();
   for (int i = 0; i < channels_; ++i)
     delete [] audio_data_[i];
 }
 
 bool AudioDevice::Start() {
   // Make sure we don't call Start() more than once.
   DCHECK_EQ(0, stream_id_);
   if (stream_id_)
     return false;
 
   AudioParameters params;
   params.format = AudioParameters::AUDIO_PCM_LOW_LATENCY;
   params.channels = channels_;
   params.sample_rate = static_cast<int>(sample_rate_);
   params.bits_per_sample = bits_per_sample_;
   params.samples_per_packet = buffer_size_;
 
-  // Ensure that the initialization task is posted on the I/O thread by
-  // accessing the I/O message loop directly. This approach avoids a race
-  // condition which could exist if the message loop of the filter was
-  // used instead.
-  DCHECK(ChildProcess::current()) << "Must be in the renderer";
-  MessageLoop* message_loop = ChildProcess::current()->io_message_loop();
-  if (!message_loop)
-    return false;
-
-  message_loop->PostTask(FROM_HERE,
+  ChildProcess::current()->io_message_loop()->PostTask(

[tommi (sloooow) - chröme, 2011/06/30 14:11:23]

should we retain the old comment about why we're accessing the io loop directly?

[henrika_dont_use, 2011/06/30 14:59:51]

Not needed since the filter now lives on the render thread and has been set up
already. The old comment was added because it was possible that the filter had
not yet been added to the IO thread when calling Start. Now, we know that this
race condition will not happen.

+      FROM_HERE,
       NewRunnableMethod(this, &AudioDevice::InitializeOnIOThread, params));
 
   return true;
 }
 
 bool AudioDevice::Stop() {
   if (!stream_id_)
     return false;
 
-  filter_->message_loop()->PostTask(FROM_HERE,
+  ChildProcess::current()->io_message_loop()->PostTask(
+      FROM_HERE,
       NewRunnableMethod(this, &AudioDevice::ShutDownOnIOThread));
 
   if (audio_thread_.get()) {
     socket_->Close();
     audio_thread_->Join();
   }
 
   return true;
 }
 
 bool AudioDevice::SetVolume(double volume) {
   if (!stream_id_)
     return false;
 
   if (volume < 0 || volume > 1.0)
     return false;
 
-  filter_->message_loop()->PostTask(FROM_HERE,
+  ChildProcess::current()->io_message_loop()->PostTask(
+      FROM_HERE,
       NewRunnableMethod(this, &AudioDevice::SetVolumeOnIOThread, volume));
 
   volume_ = volume;
 
   return true;
 }
 
 bool AudioDevice::GetVolume(double* volume) {
   if (!stream_id_)
     return false;
 
   // Return a locally cached version of the current scaling factor.
   *volume = volume_;
 
   return true;
 }
 
 void AudioDevice::InitializeOnIOThread(const AudioParameters& params) {
   stream_id_ = filter_->AddDelegate(this);
-  filter_->Send(new AudioHostMsg_CreateStream(0, stream_id_, params, true));
+  Send(new AudioHostMsg_CreateStream(stream_id_, params, true));
 }
 
 void AudioDevice::StartOnIOThread() {
   if (stream_id_)
-    filter_->Send(new AudioHostMsg_PlayStream(0, stream_id_));
+    Send(new AudioHostMsg_PlayStream(stream_id_));
 }
 
 void AudioDevice::ShutDownOnIOThread() {
   // Make sure we don't call shutdown more than once.
   if (!stream_id_)
     return;
 
-  filter_->Send(new AudioHostMsg_CloseStream(0, stream_id_));
   filter_->RemoveDelegate(stream_id_);
+  Send(new AudioHostMsg_CloseStream(stream_id_));
   stream_id_ = 0;
 }
 
 void AudioDevice::SetVolumeOnIOThread(double volume) {
   if (stream_id_)
-    filter_->Send(new AudioHostMsg_SetVolume(0, stream_id_, volume));
+    Send(new AudioHostMsg_SetVolume(stream_id_, volume));
 }
 
 void AudioDevice::OnRequestPacket(AudioBuffersState buffers_state) {
   // This method does not apply to the low-latency system.
   NOTIMPLEMENTED();
 }
 
 void AudioDevice::OnStateChanged(AudioStreamState state) {
   if (state == kAudioStreamError) {
     DLOG(WARNING) << "AudioDevice::OnStateChanged(kError)";
   }
   NOTIMPLEMENTED();
 }
 
 void AudioDevice::OnCreated(
     base::SharedMemoryHandle handle, uint32 length) {
   // Not needed in this simple implementation.
   NOTIMPLEMENTED();
 }
 
 void AudioDevice::OnLowLatencyCreated(
     base::SharedMemoryHandle handle,
     base::SyncSocket::Handle socket_handle,
     uint32 length) {
+  DCHECK(MessageLoop::current() == ChildProcess::current()->io_message_loop());
 #if defined(OS_WIN)
   DCHECK(handle);
   DCHECK(socket_handle);
 #else
   DCHECK_GE(handle.fd, 0);
   DCHECK_GE(socket_handle, 0);
 #endif
   DCHECK(length);
 
-  // TODO(crogers) : check that length is big enough for buffer_size_
-
   shared_memory_.reset(new base::SharedMemory(handle, false));
   shared_memory_->Map(length);
 
   socket_.reset(new base::SyncSocket(socket_handle));
   // Allow the client to pre-populate the buffer.
   FireRenderCallback();
 
   audio_thread_.reset(
       new base::DelegateSimpleThread(this, "renderer_audio_thread"));
   audio_thread_->Start();
 
-  if (filter_) {
-    filter_->message_loop()->PostTask(FROM_HERE,
-        NewRunnableMethod(this, &AudioDevice::StartOnIOThread));
-  }
+  MessageLoop::current()->PostTask(
+      FROM_HERE,
+      NewRunnableMethod(this, &AudioDevice::StartOnIOThread));
 }
 
 void AudioDevice::OnVolume(double volume) {
   NOTIMPLEMENTED();
 }
 
+void AudioDevice::Send(IPC::Message* message) {
+  filter_->Send(message);
+}
+
 // Our audio thread runs here.
 void AudioDevice::Run() {
-    audio_thread_->SetThreadPriority(base::kThreadPriority_RealtimeAudio);
+  audio_thread_->SetThreadPriority(base::kThreadPriority_RealtimeAudio);
 
   int pending_data;
   const int samples_per_ms = static_cast<int>(sample_rate_) / 1000;
   const int bytes_per_ms = channels_ * (bits_per_sample_ / 8) * samples_per_ms;
 
   while (sizeof(pending_data) == socket_->Receive(&pending_data,
                                                   sizeof(pending_data)) &&
                                                   pending_data >= 0) {
-    {
-      // Convert the number of pending bytes in the render buffer
-      // into milliseconds.
-      audio_delay_milliseconds_ = pending_data / bytes_per_ms;
-    }
+    // Convert the number of pending bytes in the render buffer
+    // into milliseconds.
+    audio_delay_milliseconds_ = pending_data / bytes_per_ms;
 
     FireRenderCallback();
   }
 }
 
 void AudioDevice::FireRenderCallback() {
   if (callback_) {
     // Update the audio-delay measurement then ask client to render audio.
     callback_->Render(audio_data_, buffer_size_, audio_delay_milliseconds_);
 
     // Interleave, scale, and clip to int16.
     int16* output_buffer16 = static_cast<int16*>(shared_memory_data());
     media::InterleaveFloatToInt16(audio_data_, output_buffer16, buffer_size_);
   }
 }